1. Field of the Invention
The present invention relates generally to the field of teleconferencing, and more particularly to a system and method for dynamically establishing optimum audio quality in an audio conference.
2. Background of the Invention
The telecommunications industry is constantly creating alternatives to travel for reaching a meeting forum. Teleconferencing has enabled many users to avoid long and expensive trips merely to meet with others to discuss business related topics and make important decisions. In addition, teleconferencing often replaces face to face meetings involving even the shortest of trips, such as those involving office locations relatively close in distance.
Typically, teleconferencing efficiency increases as the quality of the audio increases. Unfortunately, the quality of the audio in teleconferencing can be compromised by quality of conventional telephone lines. Telephone lines often vary markedly from one telephone line to another telephone line. Consequently, the data rates that can be achieved utilizing the telephone lines vary considerably as well. The varying telephone lines and data rates that can be achieved is particularly of concern with respect to international and/or long-distance connections, since the variation of the telephone lines and the data rates creates a potentially unreliable communication system.
Further, obtaining the best audio quality for a particular connection is complicated by the fact that the connection type (e.g., long-distance, international, etc.) is not typically known until the actual connection is established between two communication devices. In addition, when speech compressors are utilized in an attempt to improve audio quality, matching the data rate of the speech compressors must be considered as well.
Wideband audio-over-POTS (plain old telephone system) combines a modem with a codec in order to send compressed speech over a phone line. These wideband audio-over-POTS systems are often used in broadcasting to send higher-quality audio over convention telephone lines. Wideband audio-over-data (such as IP or ISDN) systems are also available and operate similarly. However, these systems are still limited by the communications line, itself, and thus cannot send audio at a faster data rate or at a higher bandwidth than the communication lines can accommodate.
Codecs (coder/decoder) compress speech into data for transmission, sometimes via conventional telephone lines. While the compression of the speech allows for a higher quality transmission of audio data, the bandwidth of the audio data is fixed by the codec. The data rate is also dependent upon the bandwidth, and thus, in these embodiments both the data rate and the bandwidth are static. For instance, G.711 provides a 3.3 kHz bandwidth codec capable of transmitting data at 64 kbps.
Alternatively, multi-rate codecs are capable of operating at different rates. In other words, multi-rate codecs provide a fixed audio bandwidth, but different quality levels depending on data rates. For instance, G.722 can provide 7 kHz audio bandwidth capable of transmitting data at either 48 kbps or 64 kbps. As another example, G.722.1 provides 7 kHz audio bandwidth and can transmit data from 24 kbps to 32 kbps. Although varying data rates are provided for each bandwidth, the audio bandwidth is static. Accordingly, data rates outside of those data rates specifically prescribed by the particular audio bandwidth cannot be achieved. Furthermore, in wideband-over-POTS systems, codecs are typically disabled when the data rate drops below a certain level, and narrowband audio is utilized instead to provide audio. Thus, codecs are not practical when acceptable audio cannot be provided due to lack of availability of a specific data rate via a conventional telephone line.
Therefore, it can be appreciated that there exists a need for a system and method for dynamically establishing optimum audio quality in an audio conference.